commit 77490b95c0ec620ff8437e7a74ac4fcf3120a232
author Karl Wiberg <kwiberg@webrtc.org> Wed Mar 21 14:18:42 2018
committer Commit Bot <commit-bot@chromium.org> Wed Mar 21 23:29:53 2018
tree b1c1e8fef8901cb03783fc0c837e53056729726f
parent ccd1523be4ee040c91be63ed612c8f2a0ce38cdf

Pass a real audio codec pair ID to encoders that we create

Bug: webrtc:8941
Change-Id: I0c0cb547e8424dd80b93e240bd0d40a9269bd1fc
Reviewed-on: https://webrtc-review.googlesource.com/63263
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22546}

audio/audio_send_stream.cc

diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 8ff34ed..71a4319 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -497,8 +497,8 @@
 
   RTC_DCHECK(new_config.encoder_factory);
   std::unique_ptr<AudioEncoder> encoder =
-      new_config.encoder_factory->MakeAudioEncoder(spec.payload_type,
-                                                   spec.format, rtc::nullopt);
+      new_config.encoder_factory->MakeAudioEncoder(
+          spec.payload_type, spec.format, new_config.codec_pair_id);
 
   if (!encoder) {
     RTC_DLOG(LS_ERROR) << "Unable to create encoder for " << spec.format;

call/audio_send_stream.h

diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h
index 908da11..f84c6cb 100644
--- a/call/audio_send_stream.h
+++ b/call/audio_send_stream.h
@@ -15,6 +15,7 @@
 #include <string>
 #include <vector>
 
+#include "api/audio_codecs/audio_codec_pair_id.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_encoder_factory.h"
 #include "api/audio_codecs/audio_format.h"
@@ -120,6 +121,7 @@
 
     rtc::Optional<SendCodecSpec> send_codec_spec;
     rtc::scoped_refptr<AudioEncoderFactory> encoder_factory;
+    rtc::Optional<AudioCodecPairId> codec_pair_id;
 
     // Track ID as specified during track creation.
     std::string track_id;

media/engine/webrtcvoiceengine.cc

diff --git a/media/engine/webrtcvoiceengine.cc b/media/engine/webrtcvoiceengine.cc
index 809ea54..04f2997 100644
--- a/media/engine/webrtcvoiceengine.cc
+++ b/media/engine/webrtcvoiceengine.cc
@@ -751,7 +751,8 @@
       const rtc::Optional<std::string>& audio_network_adaptor_config,
       webrtc::Call* call,
       webrtc::Transport* send_transport,
-      const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory)
+      const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
+      const rtc::Optional<webrtc::AudioCodecPairId> codec_pair_id)
       : call_(call),
         config_(send_transport),
         send_side_bwe_with_overhead_(
@@ -765,6 +766,7 @@
     config_.rtp.extensions = extensions;
     config_.audio_network_adaptor_config = audio_network_adaptor_config;
     config_.encoder_factory = encoder_factory;
+    config_.codec_pair_id = codec_pair_id;
     config_.track_id = track_id;
     rtp_parameters_.encodings[0].ssrc = ssrc;
 
@@ -1766,16 +1768,9 @@
   rtc::Optional<std::string> audio_network_adaptor_config =
       GetAudioNetworkAdaptorConfig(options_);
   WebRtcAudioSendStream* stream = new WebRtcAudioSendStream(
-      ssrc,
-      sp.cname,
-      sp.id,
-      send_codec_spec_,
-      send_rtp_extensions_,
-      max_send_bitrate_bps_,
-      audio_network_adaptor_config,
-      call_,
-      this,
-      engine()->encoder_factory_);
+      ssrc, sp.cname, sp.id, send_codec_spec_, send_rtp_extensions_,
+      max_send_bitrate_bps_, audio_network_adaptor_config, call_, this,
+      engine()->encoder_factory_, codec_pair_id_);
   send_streams_.insert(std::make_pair(ssrc, stream));
 
   // At this point the stream's local SSRC has been updated. If it is the first